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gerrit-public.fairphone.software / platform / external / webrtc / 98f53510b222f71fdd8b799b2f33737ceeb28c61
  1. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  2. ebc0b4e Use webrtc/base/logging.h for rtp_rtcp. by Peter Boström · 9 years ago
  3. 605db69 Disable EndToEndTest.AssignsTrans... for memcheck by henrik.lundin · 9 years ago
  4. 6408174 Fix for "Android audio playout doesn't support non-call media stream" by henrika · 9 years ago
  5. 83585c9 VideoCapturerAndroid: More frequent and verbose logging by magjed · 9 years ago
  6. ec9d187 Added override keyword to overridden methods to stop compiler warnings. by rlester · 9 years ago
  7. fce4a94 RentACodec: New class that takes over part of ACMCodecDB's job by kwiberg · 9 years ago
  8. 77d0d6e When all connections timed out on writing, delete them all. BUG=5111 by honghaiz · 9 years ago
  9. f116bd0 Call OnSentPacket for all packets sent in the test framework. by stefan · 9 years ago
  10. f1dcd46 UBSan: Add blacklist files for WebRTC standalone. by Henrik Kjellander · 9 years ago
  11. 9397d84 Roll chromium_revision 625f6c8..657e8d9 (356202:356260) by kjellander · 9 years ago
  12. 27f6fd3 Remove noparent from talk/OWNERS. by pbos · 9 years ago
  13. 5ddee02 Landmine: clobber to remove out/{Debug,Release}/args.gn by Henrik Kjellander · 9 years ago
  14. 4f847da Use webrtc/base/checks.h in desktop_capture. by pbos · 9 years ago
  15. 85a0496 Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  16. 2a0a2a4 Add stats for used video codec type for a sent video stream: by asapersson · 9 years ago
  17. 18ba3e2 Roll chromium_revision faa5502..625f6c8 (356073:356202) by kjellander · 9 years ago
  18. 18a944b Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ ) by deadbeef · 9 years ago
  19. d3b26d9 Adding the ability to change ICE servers through SetConfiguration. by deadbeef · 9 years ago
  20. 2b55867 Exposing DTLS transport state from TransportChannel. by deadbeef · 9 years ago
  21. b0bb77f Reland of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1416773003/ ) by guoweis · 9 years ago
  22. 8f46c63 Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) by deadbeef · 9 years ago
  23. aed571f Roll chromium_revision 27af50f..faa5502 (356022:356073) by kjellander · 9 years ago
  24. e2a83ee Introduce rtc::ArrayView<T>, which keeps track of an array that it doesn't own by Karl Wiberg · 9 years ago
  25. ac9d92c Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  26. 4cba4eb Disable denoising for VP9 by default. by pbos · 9 years ago
  27. 65e7d4c Remove CanCreateAndDestroyManyVideoStreams. by Peter Boström · 9 years ago
  28. c4ef143 Revert "Add GN Build file for rtc_sound target." by Henrik Kjellander · 9 years ago
  29. 717432f Remove network_enabled_crit_ in call.cc. by mflodman · 9 years ago
  30. 09b38f3 Re-enable VP9 resize test. by Marco · 9 years ago
  31. 7ef0553 Fix for Win GN Build. by tfarina · 9 years ago
  32. 2d3747d Fix for Mac GN BUILD. by tfarina · 9 years ago
  33. e9eca8f Removing AudioCoding class, a.k.a the new ACM API by henrik.lundin · 9 years ago
  34. f054819 Add GN Build file for rtc_sound target. by tfarina · 9 years ago
  35. 213b598 Roll chromium_revision c86a4e2..27af50f (356002:356022) by kjellander · 9 years ago
  36. 415d2cd Use webrtc/base/logging.h for video. by Peter Boström · 9 years ago
  37. f9af108 Roll chromium_revision c708f39..c86a4e2 (355993:356002) by kjellander · 9 years ago
  38. 484e548 Roll chromium_revision bbfaf80..c708f39 (355989:355993) by kjellander · 9 years ago
  39. eb2a91e Roll chromium_revision 5512fa0..bbfaf80 (355985:355989) by kjellander · 9 years ago
  40. 7542ed6 Roll chromium_revision da8662f..5512fa0 (355980:355985) by kjellander · 9 years ago
  41. 9ec27e1 Roll chromium_revision da9833c..da8662f (355969:355980) by kjellander · 9 years ago
  42. 5d9b92b Update Bind to match its comments and always capture by value. Also update the generated count to 9 args. by noahric · 9 years ago
  43. 2dd8bf8 Roll chromium_revision 53f0e22..da9833c (355953:355969) by kjellander · 9 years ago
  44. 7d35afd Roll chromium_revision bd99556..53f0e22 (355580:355953) by kjellander · 9 years ago
  45. 401fb06 SurfaceTextureHelper: Remove use of quitSafely() because it's API lvl 18 by magjed · 9 years ago
  46. 238b15d SurfaceViewRenderer: Remove use of quitSafely() because it's API lvl 18 by magjed · 9 years ago
  47. c3402fc EGL10.eglCreateWindowSurface(): Replace Surface input with SurfaceHolder by magjed · 9 years ago
  48. 90d67dd Remove two more deprecated methods from SocketAddress API. by tfarina · 9 years ago
  49. 49e196a Remove VideoFrameType aliases for FrameType. by Peter Boström · 9 years ago
  50. a99069d Fix win32 header include order in rtp_utility.h. by pbos · 9 years ago
  51. 225789d Move logging CriticalSection into implementation. by Peter Boström · 9 years ago
  52. aa04299 Don't wait until distant future to shut down video app. by mflodman · 9 years ago
  53. 27dfe20 Remove final from rtc::Buffer. by noahric · 9 years ago
  54. 1e737c6 Fix thread safety in VcmCapturer. by Peter Boström · 9 years ago
  55. bbe876f Set send times in send time history via OnSentPacket. by stefan · 9 years ago
  56. 9a4cd87 Add support for handling reordered SS data on the receive-side for VP9. by asapersson · 9 years ago
  57. a3587fb clean up field_trial_default target, to be used by remoting_perftests. by guoweis · 9 years ago
  58. 00507f8 Separate StunProber::Start into Prepare and Run so we could create multiple of them and send out STUN pings at regular interval. by guoweis · 9 years ago
  59. 4f6a8b5 Revert of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1406153005/ ) by guoweis · 9 years ago
  60. e26ce1b Reland of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1423443002/ ) by guoweis · 9 years ago
  61. 8c425aa Android: Replace EGL14 with EGL10 by magjed · 9 years ago
  62. ff134eb talk: Use NDEBUG macro. by tfarina · 9 years ago
  63. c80741f Fixing some issues with the direction attribute of m-lines in offers. by deadbeef · 9 years ago
  64. 56149e5 Roll chromium_revision 7c002e5..bd99556 (355518:355580) by kjellander · 9 years ago
  65. b7edb88 Prevent BWE rampdowns without new loss reports. by pbos · 9 years ago
  66. 4a859fd Roll chromium_revision 2c4120b..7c002e5 (355476:355518) by kjellander · 9 years ago
  67. 797ef12 Added StopAecDump function to PeerConnectionFactory. by ivoc · 9 years ago
  68. a74c08d Move i420 files to the right location by Henrik Kjellander · 9 years ago
  69. 48e66b4 GN: Remove build_overrides/v8.gni by Henrik Kjellander · 9 years ago
  70. 4f4ec0a Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  71. b1ce663 Allow encoders to fall back dynamically to software. by noahric · 9 years ago
  72. b788bc2 Add Mac-specific resource to modules_unittests.isolate by Henrik Kjellander · 9 years ago
  73. 93ea78b Add test resources to libjingle_media_unittest.isolate by Henrik Kjellander · 9 years ago
  74. 9589e2a Update isolate files for swarming tests by Henrik Kjellander · 9 years ago
  75. 4f47ed4 Roll chromium_revision fecea52..2c4120b (355266:355476) by kjellander · 9 years ago
  76. 522fac7 Roll chromium_revision 9a72f7c..fecea52 (355218:355266) by kjellander · 9 years ago
  77. affa39c Remove time constraint on first retransmit of a packet. by sprang · 9 years ago
  78. c96df77 - Introduce internal classes WebRtcAudio[Send|Receive]Stream in WebRtcVoiceMediaChannel. by solenberg · 9 years ago
  79. f4d23b2 Remove MockVideoCapturer by magjed · 9 years ago
  80. dfa2815 Update receive report SSRCs on RemoveSendStream. by Peter Boström · 9 years ago
  81. 0c478b3 Rename ChannelGroup to CongestionController and move to webrtc/call/. by mflodman · 9 years ago
  82. edcbd56 Adding the OnePlus 2 device to AEC and NS blacklists. by henrika · 9 years ago
  83. 0a87ffc Fix bug in how send timestamps are converted to 24 bits. by Stefan Holmer · 9 years ago
  84. e378702 ChannelGroup cleanup. by mflodman · 9 years ago
  85. 45c136b Adds support for Bluetooth headsets to the iOS audio layer. by henrika · 9 years ago
  86. 6e58720 Introduce rtc::Maybe<T>, which either contains a T or not. by Karl Wiberg · 9 years ago
  87. b64a32b Remove old VideoFrame::Reset. by Peter Boström · 9 years ago
  88. 3b7c793 New DtlsIdentityStoreInterface::RequestIdentity added that takes rtc::KeyParams. The old RequestIdentity still exists that take rtc::KeyType. by hbos · 9 years ago
  89. a01d440 Revert of Add experiment on weak ping delay during call set up time (patchset #4 id:60001 of https://codereview.webrtc.org/1411883002/ ) by tommi · 9 years ago
  90. 86b0160 Add stats for average QP per frame for VP8 (for received video streams): by asapersson · 9 years ago
  91. 47dcb23 Roll chromium_revision 01cbe8a..9a72f7c (355025:355218) by kjellander · 9 years ago
  92. fcab1cd Disable VP9 resize test for now. by Marco · 9 years ago
  93. e4f9650 Remove system_wrappers/interface/trace_event.h by tommi · 9 years ago
  94. 3cf20ed Will re-enable after libvpx roll, needs to be updated. by Marco · 9 years ago
  95. 0a617e2 Remove the default send channel in WVoE. by solenberg · 9 years ago
  96. 3bf69b1 Add experiment on weak ping delay during call set up time by Guo-wei Shieh · 9 years ago
  97. 30a5b5e passing |buffer| by reference in AndroidVideoCapturer::OnIncomingFrame by olka · 9 years ago
  98. 3866c4f Testing that waiting for a condition variable waits. by hta · 9 years ago
  99. 43e83d4 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
  100. 5a197dd Remove files added by mistake. by Fredrik Solenberg · 9 years ago